I have a customer who has just put a PolyCom 330 in a home office that is on the end of a WildBlue.Com satellite link.
The phone registers fine and Asterisk reports a 1919ms latency via the SIP SHOW PEERS command. I know this is not optimal but the phone registers and she can receive and make phone calls.
The problem is that while she can hear callers PERFECTLY, callers can’t hear her. The downstream audio is crystal clear but the upstream audio is almost non-existant with very little other than the occasional word or syllable getting through but mostly dead air.
I’ve allowed only G729 and her Satellite Upload speed is currently running at 90k on average. Download is around 280k according to DSLReports. I tried allow=g729:30 based upon something I read about g279 packetization but sip reload kicked this out as invalid so i change it back to just allow=g729.
Any suggestions for things I can change/adapt/try to make this work across her Satellite link?
I know i’m in for some latency in 2way conversations but the fact that she can hear perfectly means that the RTP stream is at least making the round trip.